[Asterisk-Users] Need to block incoming collect calls
Hi everybody,, I need to block incoming collect calls to my Asterisk box but I could not find out where to do that. Went to zaptel.h but I did not see any timing which can be applied to collect calls. Does anybody knows if I can set this up in Asterisk? I'm using an E100P connected to the PSTN and a T100P connected to a Zhone 100. Version: Asterisk CVS-05/30/04-16:28:04 thank you Oz _ MSN Messenger: instale grátis e converse com seus amigos. http://messenger.msn.com.br ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need to block incoming collect calls
All right Steve. I'll ask them.. But if anybody knows that, please post an answer to the list. This is a very important Asterisk security configuration to avoid people call you without having to pay the call.. thank you Oz From: Steve Totaro [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Need to block incoming collect calls Date: Sat, 24 Jul 2004 11:57:05 -0400 I dont know about blocking in * but you should be able give the telco a call and tell them no collect calls. - Original Message - From: Osvaldo Mundim Junior [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, July 24, 2004 10:06 AM Subject: [Asterisk-Users] Need to block incoming collect calls Hi everybody,, I need to block incoming collect calls to my Asterisk box but I could not find out where to do that. Went to zaptel.h but I did not see any timing which can be applied to collect calls. Does anybody knows if I can set this up in Asterisk? I'm using an E100P connected to the PSTN and a T100P connected to a Zhone 100. Version: Asterisk CVS-05/30/04-16:28:04 thank you Oz _ MSN Messenger: instale grátis e converse com seus amigos. http://messenger.msn.com.br ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ MSN Messenger: instale grátis e converse com seus amigos. http://messenger.msn.com.br ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] collect calls
Is it possible to set in Asterisk? Not to accept collect calls? Oz From: Osvaldo Mundim Junior [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] collect calls Date: Mon, 19 Jul 2004 16:33:19 -0300 Hi, Does anybody knows where can I change timing for collect calls? tks Oz _ MSN Messenger: instale grátis e converse com seus amigos. http://messenger.msn.com.br ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ MSN Messenger: instale grátis e converse com seus amigos. http://messenger.msn.com.br ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] collect calls
Hi, Does anybody knows where can I change timing for collect calls? tks Oz _ MSN Messenger: instale grátis e converse com seus amigos. http://messenger.msn.com.br ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] do_monitor: Bad file descriptor
Did anybody get this error message before: chan_zap.c:5044 do_monitor: select return -1: Bad file descriptor When it's happening, Asterisk gets freezed and talkers can not hear each other. This message appears like in a loop at the server's screen. thank you Oz _ MSN Messenger: instale grátis e converse com seus amigos. http://messenger.msn.com.br ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ATA call
Hey all! I'm having problems trying to set up an ATA 186 with my Asterisk box. When I get the phone to place the call, I type the extension and I only get busy signal after 5 seconds. So I can't call my Asterisk box from my ATA and either call from my Asterisk to my ATA. Does anybody know what can be happing? Log is attached.. tks regards Oz 8 headers, 0 lines Retransmitting #1 (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.150:5060;received=200.167.103.219 From: sip:[EMAIL PROTECTED];tag=3346186142 To: sip:[EMAIL PROTECTED];user=phone;tag=as36ac1b92 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm=asterisk, nonce=4bd7a841 Content-Length: 0 290Ñ to 200.167.103.219:1025 Sip read: LI INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.150:5060 From: sip:[EMAIL PROTECTED];tag=3346186142 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE Contact: sip:[EMAIL PROTECTED]:5060;transport=udp User-Agent: Cisco ATA 186 v2.16.1 ata18x (030709a) Proxy-Authorization: Digest username=porto,realm=asterisk,nonce=4bd7a841,uri=sip:[EMAIL PROTECTED] .77,response=1ecb99d4d5e23be179a9eb55eb33c62a Expires: 300 Content-Length: 250 Content-Type: application/sdp v=0 o=porto 3642 3642 IN IP4 192.168.0.150 s=ATA186 Call c=IN IP4 192.168.0.150 t=0 0 m=audio 16384 RTP/AVP 18 8 0 101 a=rtpmap:18 G729/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 12 headers, 11 lines Using latest request as basis request Sending to 192.168.0.150 : 5060 (NAT) Found audio format UNKN Found audio format ALAW Found audio format UNKN Found audio format UNKN Found description format G729 Found description format PCMA Found description format PCMU Found description format telephone-event Capabilities: us - 256, them - 268/0, combined - 256 Non-codec capabilities: us - 1, them - 1, combined - 1 10 headers, 0 lines Reliably Transmitting: OPTIONS sip:200.167.103.219:1025 SIP/2.0 Via: SIP/2.0/UDP 200.170.156.77:5060;branch=z9hG4bK1937468f From: asterisk sip:[EMAIL PROTECTED];tag=as5566fcc8 To: sip:200.167.103.219:1025 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 200.167.103.219:1025 Sip read: LI ACK sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.150:5060;received=200.167.103.219 From: sip:[EMAIL PROTECTED];tag=3346186142 To: sip:[EMAIL PROTECTED];user=phone;tag=as36ac1b92 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK User-Agent: Cisco ATA 186 v2.16.1 ata18x (030709a) Content-Length: 0 8 headers, 0 lines Sip read: LI INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.150:5060 From: sip:[EMAIL PROTECTED];tag=3346186142 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE Contact: sip:[EMAIL PROTECTED]:5060;transport=udp User-Agent: Cisco ATA 186 v2.16.1 ata18x (030709a) Proxy-Authorization: Digest username=porto,realm=asterisk,nonce=514a024a,uri=sip:[EMAIL PROTECTED] .77,response=adb7da64c3f557d1db20b699c04f6d84 Expires: 300 Content-Length: 250 Content-Type: application/sdp v=0 o=porto 3692 3692 IN IP4 192.168.0.150 s=ATA186 Call c=IN IP4 192.168.0.150 t=0 0 m=audio 16384 RTP/AVP 18 8 0 101 a=rtpmap:18 G729/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 12 headers, 11 lines Using latest request as basis request Sending to 192.168.0.150 : 5060 (non-NAT) Found audio format UNKN Found audio format ALAW Found audio format UNKN Found audio format UNKN Found description format G729 Found description format PCMA Found description format PCMU Found description format telephone-event Capabilities: us - 256, them - 268/0, combined - 256 Non-codec capabilities: us - 1, them - 1, combined - 1 Reliably Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.150:5060;received=200.167.103.219 From: sip:[EMAIL PROTECTED];tag=3346186142 To: sip:[EMAIL PROTECTED];user=phone;tag=as046b1041 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm=asterisk, nonce=6512ffab Content-Length: 0 to 200.167.103.219:1025 Sip read: LI ACK sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.150:5060;received=200.167.103.219 From: sip:[EMAIL PROTECTED];tag=3346186142 To: sip:[EMAIL PROTECTED];user=phone;tag=as36ac1b92 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK User-Agent: Cisco ATA 186 v2.16.1 ata18x (030709a) Content-Length: 0 8 headers, 0 lines Retransmitting #1 (no NAT): OPTIONS
Re: [Asterisk-Users] ATA call
Hi Doug, I do use the SIP 2.16x on my ATA 186. But I can not see the IP address of my ATA on show sip peers. What I can see is: - Original Message - From: Doug Shubert [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, January 06, 2004 9:09 AM Subject: Re: [Asterisk-Users] ATA call Is your ATA running SIP if so, what version (2.16?) With SIP, then * extensions.conf and sip.conf files are configured you should see the following asterisk3*CLI sip show peers Name/usernameHost Mask Port Status 3000/300010.0.0.30 (D) 255.255.255.255 5060 OK (15 ms) 9000/900010.0.0.90 (D) 255.255.255.255 5060 OK (47 ms) ext 3000 is the Cisco ATA 186 and ext 9000 is the Cisco 7960 to test an extension from the CLI CLIdial ext. # you should hear your ATA ring Doug Osvaldo Mundim Junior wrote: Hey all! I'm having problems trying to set up an ATA 186 with my Asterisk box. When I get the phone to place the call, I type the extension and I only get busy signal after 5 seconds. So I can't call my Asterisk box from my ATA and either call from my Asterisk to my ATA. Does anybody know what can be happing? Log is attached.. tks regards Oz Name: ast_log.txt ast_log.txtType: Plain Text (text/plain) Encoding: quoted-printable -- FREE Unlimited Worldwide Voip calling set-up an account and start saving today! http://www.voippages.com ext. 7000 http://www.pulver.com/fwd/ ext. 83740 free IP phone software @ http://www.xten.com/ http://iaxclient.sourceforge.net/iaxcomm/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA call
Hi Doug, I do use the SIP 2.16x on my ATA 186. But I can not see the IP address of my ATA on show sip peers. What I can see is: Name/usernameHost Mask Port Status porto/porto (Unspecified) (D) 255.255.255.255 0UNKNOWN Just one thing which I did not mention on the last email is that my ATA is behing NAT. Oz - Original Message - From: Doug Shubert [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, January 06, 2004 9:09 AM Subject: Re: [Asterisk-Users] ATA call Is your ATA running SIP if so, what version (2.16?) With SIP, then * extensions.conf and sip.conf files are configured you should see the following asterisk3*CLI sip show peers Name/usernameHost Mask Port Status 3000/300010.0.0.30 (D) 255.255.255.255 5060 OK (15 ms) 9000/900010.0.0.90 (D) 255.255.255.255 5060 OK (47 ms) ext 3000 is the Cisco ATA 186 and ext 9000 is the Cisco 7960 to test an extension from the CLI CLIdial ext. # you should hear your ATA ring Doug Osvaldo Mundim Junior wrote: Hey all! I'm having problems trying to set up an ATA 186 with my Asterisk box. When I get the phone to place the call, I type the extension and I only get busy signal after 5 seconds. So I can't call my Asterisk box from my ATA and either call from my Asterisk to my ATA. Does anybody know what can be happing? Log is attached.. tks regards Oz Name: ast_log.txt ast_log.txtType: Plain Text (text/plain) Encoding: quoted-printable -- FREE Unlimited Worldwide Voip calling set-up an account and start saving today! http://www.voippages.com ext. 7000 http://www.pulver.com/fwd/ ext. 83740 free IP phone software @ http://www.xten.com/ http://iaxclient.sourceforge.net/iaxcomm/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA call
Some times the sip show peers shows me: Name/usernameHost Mask Port Status porto/porto (Unspecified) (D) 255.255.255.255 0UNKNOWN and some times shows me: Name/usernameHost Mask Port Status porto/porto 200.167.103.219 (D) 255.255.255.255 1025 LAGGED (815 ms) Does the port supposed to be 5060? Oz - Original Message - From: Doug Shubert [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, January 06, 2004 9:09 AM Subject: Re: [Asterisk-Users] ATA call Is your ATA running SIP if so, what version (2.16?) With SIP, then * extensions.conf and sip.conf files are configured you should see the following asterisk3*CLI sip show peers Name/usernameHost Mask Port Status 3000/300010.0.0.30 (D) 255.255.255.255 5060 OK (15 ms) 9000/900010.0.0.90 (D) 255.255.255.255 5060 OK (47 ms) ext 3000 is the Cisco ATA 186 and ext 9000 is the Cisco 7960 to test an extension from the CLI CLIdial ext. # you should hear your ATA ring Doug Osvaldo Mundim Junior wrote: Hey all! I'm having problems trying to set up an ATA 186 with my Asterisk box. When I get the phone to place the call, I type the extension and I only get busy signal after 5 seconds. So I can't call my Asterisk box from my ATA and either call from my Asterisk to my ATA. Does anybody know what can be happing? Log is attached.. tks regards Oz Name: ast_log.txt ast_log.txtType: Plain Text (text/plain) Encoding: quoted-printable -- FREE Unlimited Worldwide Voip calling set-up an account and start saving today! http://www.voippages.com ext. 7000 http://www.pulver.com/fwd/ ext. 83740 free IP phone software @ http://www.xten.com/ http://iaxclient.sourceforge.net/iaxcomm/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E1 in Brazil
Hi, I've asked them about the switch and they told me that its a Siemens EWSD.. Regards Oz On 9/25/03 11:32 AM, Ing. Angel Gomez Garcia [EMAIL PROTECTED] wrote: Hi. Do you know what switch your telco has ? The one they are using to provide you the service. Osvaldo Mundim Junior wrote: ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E1 in Brazil
I'm sorry, but I don¹t know. I'll ask them and I let you know. Oz On 9/25/03 11:32 AM, Ing. Angel Gomez Garcia [EMAIL PROTECTED] wrote: Hi. Do you know what switch your telco has ? The one they are using to provide you the service. Osvaldo Mundim Junior wrote: Hey all! I had an experience trying to set up an E1 in Brazil which could help somebody. In Brazil is very common telcos to have just R2 digital as their primary signaling. As I were trying to set up an E100P, which does not support R2 yet, I had to test an other signaling which works perfectly with Asterisk. They call this signaling as RDSI, using ccs as framing and PA (primary access) as coding. This RDSI are 30 channels completely digital which uses 128k per channel (2Mb). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E1 in Brazil
Yes, it does. Eduardo was right. They call RDSI as ISDN in Brazil. And its working with an E100P. regards Oz On 9/25/03 11:59 AM, Andrew Kohlsmith [EMAIL PROTECTED] wrote: RDSI and ISDN are the same thing. RDSI is ISDN said in portuguese. The E100P does not do ISDN, does it? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call volume on ATA 188
Hi all! Guys, I'm testing the Cisco ATA 188 with my Asterisk (version 09/10/03) and I faced with a low call volume heard just for people who is not under the ATA. I mean, if I call a person whose extension is connected at the ATA, he can hear me perfectly, but I get a low call volume. Is it possible to change this call volume? Can I do something in order to get a little high volume on my side? Tks in advance! Oz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FW: Call volume on ATA 188
Hi all! Guys, I'm testing the Cisco ATA 188 with my Asterisk (version 09/10/03) and I faced with a low call volume heard just for people who is not under the ATA. I mean, if I call a person whose extension is connected at the ATA, he can hear me perfectly, but I get a low call volume. Is it possible to change this call volume? Can I do something in order to get a little high volume on my side? Tks in advance! Oz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
